[Freeswitch-users] FS6947 - Tuning Opus bandwidth

Sat Oct 15 00:16:13 MSD 2016

Since when? I don't see it in FS 1.6.10.

Emrah
> On Oct 14, 2016, at 6:50 PM, Brian West <brian at freeswitch.org> wrote:
> 
> We do have 12khz and 24khz I added the specifically when we added SILK, we can resample to and from both rates. :)
> 
> On Fri, Oct 14, 2016 at 7:27 AM, Dragos Oancea <dragos.oancea at athonet.com <mailto:dragos.oancea at athonet.com>> wrote:
> Hi
> 
> Why would you say opus at 8000@20i does not sound very good ? What do you
> mean exactly ? Your call quality is bad or you mean you don't like the
> fact that the sampling rate is low ?
> 
> You can play with the sampling rates by setting maxplaybackrate and
> sprop_maxcapurerate and enabling asymmetric_samplerates (which is an
> experimental feature so far) in your opus.conf.xml  .
> 
> FS does not have 12 khz or 24 khz because they are not much used for
> Voip. As for opus @ 16 khz perhaps it will be added.
> 
> But if you use opus @ 48 khz and you just change maxaveragebitrate and
> maxplaybackrate accordingly you should get only WIDEBAND from the
> encoder anyway .
> The decoder should decode at any sample rate.
> We're working on a document (sort of manual) for the Opus module and
> hopefully it will be released soon.
> 
> As for the issue why codec settings cannot be set from FS's dialplan , I
> think its a missing feature that affects other audio codecs too and I
> think it would be very useful.
> 
> We needed opus at 8000hz for transcoding and I tried to explain some things
> here (see my comment at the bottom of the page ):
> https://freeswitch.org/confluence/display/FREESWITCH/mod_opus <https://freeswitch.org/confluence/display/FREESWITCH/mod_opus>
> 
> 
> Basically if you do heavy transcoding to PCMA / PCMU which is 8000 hz
> you'll want to avoid resampling 48 khz <-> 8 khz  - we did tests and by
> avoiding resampling we were saving 20-30 % CPU .
> 
> Regards,
> Dragos
> 
> 
> 
> On 14/10/2016 14:02, Emrah wrote:
> > Hi there,
> > Revisiting this issue. I see that I can set my Opus codec with
> > Opus at 8000@20i, but 8khz seems to be the only alternative profile I can
> > use. I see only 2 extreme options when the module is loaded or unloaded
> > that it's either 48khz, mono or stereo and packet size, or 8khz, mono or
> > stereo and packet size. Can someone clarify why there is nothing in
> > between? And what exactly this setting does? Opus at 8000h@20i definitely
> > doesn't sound very good. I'd rather have a compromise for tough network
> > conditions.
> > Since these are parameters I can dynamically set on my dialplan, the
> > question then becomes why can't I fully manipulate my Opus stack from
> > the dialplan?
> >
> > Thanks!
> >> On Jun 2, 2015, at 6:23 AM, Emrah <lists at kavun.ch <mailto:lists at kavun.ch>
> >> <mailto:lists at kavun.ch <mailto:lists at kavun.ch>>> wrote:
> >>
> >> Hi there,
> >> @Mike: yes, but in a commonsensical approach the Opus library on the
> >> client's side would resample and therefore optimize the codec and the
> >> bandwidth accordingly up to FS.
> >> @Julien, I saw the setting for Opus globally, but it defeats the
> >> purpose. I don't want to limit the bandwidth of Opus for all
> >> instances. I'd like to optimize Opus on a per call basis.
> >>
> >> Thanks for the replies
> >>> On May 31, 2015, at 2:13 PM, Michael Jerris <mike at jerris.com <mailto:mike at jerris.com>
> >>> <mailto:mike at jerris.com <mailto:mike at jerris.com>>> wrote:
> >>>
> >>> Side note, opening at the different rate I believe just makes the
> >>> opus library do the re sampling instead of FreeSWITCH.
> >>>
> >>> On Sunday, May 31, 2015, Julien Chavanton <jchavanton at gmail.com <mailto:jchavanton at gmail.com>
> >>> <mailto:jchavanton at gmail.com <mailto:jchavanton at gmail.com>>> wrote:
> >>>
> >>>     Hi Emrah,
> >>>
> >>>     The settings exist but they are not available from the dialplan,
> >>>     right now they can only be set globally .
> >>>     https://freeswitch.org/confluence/display/FREESWITCH/mod_opus <https://freeswitch.org/confluence/display/FREESWITCH/mod_opus>
> >>>
> >>>     You can control the bandwidth using maxplaybackrate and
> >>>     maxplaybackrate this will control the local encoder and also adds
> >>>     the corresponding FMTP parameters to the SDP to be used by the
> >>>     remote encoder (if it does implement the following draft, the
> >>>     draft is evolving but I think it as not changed)
> >>>
> >>>     https://tools <https://tools/>
> >>>     <https://tools/ <https://tools/>>.ietf.org/html/draft-ietf-payload-rtp-opus-11 <http://ietf.org/html/draft-ietf-payload-rtp-opus-11>
> >>>
> >>>     Maybe something like :
> >>>
> >>>     maxaveragebitrate 24000
> >>>     maxplaybackrate 8000
> >>>
> >>>     The discussion was getting slightly more complicated when we
> >>>     where discussing about unnecessary resampling this was not a
> >>>     problem but it was just adding extra load on the server.
> >>>
> >>>     On Sun, May 31, 2015 at 6:09 AM, Emrah <lists at kavun.ch <mailto:lists at kavun.ch>
> >>>     <javascript:_e(%7B%7D,'cvml','lists at kavun.ch <mailto:lists at kavun.ch>');>> wrote:
> >>>
> >>>         Hi list,
> >>>
> >>>         I re-read FS6947 and don't understand how this problematic
> >>>         was addressed and the issue fixed.
> >>>         The scope is simple. There should be a setting in the
> >>>         dialplan that allows downsampling of Opus for applications
> >>>         that do not require the 48khz / 2 channels framework. I.e.:
> >>>         terminating to the PSTN with Opus to take advantage of low
> >>>         bandwidth and great PLC.
> >>>         There seems to be a lot of confusion around bandwidth in
> >>>         general there. It doesn't matter if the internal clock of the
> >>>         device is always sampling at 48khz / 2ch. There are settings
> >>>         that can facilitate a lower bandwidth consumption for
> >>>         particular use cases, and it seems the reason it is not being
> >>>         implemented in FS is just a matter of being confused about
> >>>         the intent of the 48khz 2ch base.
> >>>         Please revisit this issue. FS should allow tuning of Opus
> >>>         audio / network bandwidth in the dialplan. It would optimize
> >>>         greatly lots of use cases.
> >>>         If I'm calling the PSTN, I'd rather have my client downsample
> >>>         and stream at a lower bandwidth, even if my audio capture
> >>>         would still be at 48khz / 2ch as per the RFC, and save on
> >>>         bandwidth, than transcode the full 48khz spectrum into PCM on
> >>>         my FS and minimize processing power on the client's side.
> >>>
> >>>         Jira here: https://freeswitch.org/jira/browse/FS-6947 <https://freeswitch.org/jira/browse/FS-6947>
> >>>
> >>>         Emrah
> >>>         _________________________________________________________________________
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> >
> >
> >
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